The present invention is directed to data transmission of compressed voice packets. More particularly, the present invention is directed to a telecommunication network that reduces tandeming of compressed voice packets.
Public telephone carriers offer many digital services to which customers can subscribe. Some digital services allow the customer to multiplex multiple telecommunication applications (e.g., a PBX and computer data equipment) located at the customer's premise onto a single access circuit.
One example of a digital service is a T1 circuit. A T1 circuit includes multiple frames, with each frame including twenty-four time slots, and each time slot including eight bits of information. A T1 circuit utilizes synchronous time division multiplexing ("TDM") to multiplex together information from multiple telecommunication applications.
Another example of a digital service is an Asynchronous Transfer Mode ("ATM") circuit. All information in ATM is carried in the form of fixed-length data units called "cells." ATM utilizes a form of multiplexing known as statistical multiplexing ("STM"). With STM, bandwidth is shared among all telecommunication applications, and bandwidth is used by an application only when needed.
Digital services can be used more efficiently if the voice is compressed using one of many known compression techniques. For example, uncompressed voice typically is digitized at a rate of 64 Kbps. Using known methods (e.g., International Telecommunication Union ("ITU") standard G.729 (CS-ACELP) or ITU standard G.723.1 (MP-MLQ)), voice can be compressed to a rate of eight Kbps or below, with very good quality.
Compressed packetized voice can be used in most digital services. For example, compressed packetized voice can be carried over an ATM virtual connection ("VC") as mini-packets using an ATM Adaption Layer-2 ("AAL2") adaption layer. The AAL2 adaption layer is promulgated by ITU standard I.363.2.
Although compressed packetized voice offers many advantages, it does have some disadvantages. For examples, more delays are introduced when using compressed packets. Further, when compressed voice packets undergo a series of compression and decompression, referred to as "tandeming", voice quality rapidly degrades. This is because most voice compression algorithms are not lossless, i.e., the decompressed output speech is not the same as the input speech because some errors are introduced. The errors compound with successive compressions.
FIG. 1 illustrates a telecommunication network in which compressed voice packets are subject to tandeming. In FIG. 1, a compressed voice call takes place between two locations 12 and 14 that both subscribe to the same compressed voice digital access service (i.e., ATM). Each location 12, 14 includes end user equipment 16, 30, and an interworking function ("IWF") 18, 28. An IWF is a device that provides integrated access and voice compression/decompression. In a packet network environment, the IWF typical performs the packet relay function (i.e., the forwarding of packets). IWFs that perform this function are also referred to as a "packet relay switches."
The call between locations 12 and 14 is placed over the Public Switched Telephone Network ("PSTN") 10. PSTN 10 includes a plurality of switches, including switches 22 and 24. Further, because PSTN 10 can only switch uncompressed voice packets, IWFs 20, 26 are placed at the ingress and egress of network 10. IWFs 20, 26 decompress voice packets before the packets are received by switches 22, 24, and compress voice packets before the packets are received by a receiving location.
The call from location 12 to location 14 undergoes two tandems (i.e., compression/decompression): the ingress circuit from location 12 to switch 22 (i.e., compression at IWF 18, decompression at IWF 20); and the egress circuit from switch 24 to location 14 (i.e., compression at IWF 26, decompression at IWF 28). The multiple tandeming results in a degraded voice quality.
In other situations more than two tandems can occur. For example, if a user at location A hosts an audio conference using an audio bridge with users at location B and location C, speech from location B to location C will go through four tandems. Further, if a user at location B leaves an audio voice message at location A which is later retrieved at location C, the original message will go through four tandems.
Because tandeming of compressed voice packets results in a substantial degradation of voice quality, it is desirable to have a telecommunication network that reduces the number of tandems.